libavcodec/mpc.c

   1 /*
   2  * Musepack decoder core
   3  * Copyright (c) 2006 Konstantin Shishkov
   4  *
   5  * This file is part of FFmpeg.
   6  *
   7  * FFmpeg is free software; you can redistribute it and/or
   8  * modify it under the terms of the GNU Lesser General Public
   9  * License as published by the Free Software Foundation; either
  10  * version 2.1 of the License, or (at your option) any later version.
  11  *
  12  * FFmpeg is distributed in the hope that it will be useful,
  13  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  14  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  15  * Lesser General Public License for more details.
  16  *
  17  * You should have received a copy of the GNU Lesser General Public
  18  * License along with FFmpeg; if not, write to the Free Software
  19  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  20  */
  21
  22 /**
  23  * @file libavcodec/mpc.c Musepack decoder core
  24  * MPEG Audio Layer 1/2 -like codec with frames of 1152 samples
  25  * divided into 32 subbands.
  26  */
  27
  28 #include "avcodec.h"
  29 #include "get_bits.h"
  30 #include "dsputil.h"
  31 #include "mpegaudio.h"
  32
  33 #include "mpc.h"
  34 #include "mpcdata.h"
  35
  36 static DECLARE_ALIGNED_16(MPA_INT, mpa_window[512]);
  37
  38 void ff_mpc_init(void)
  39 {
  40     ff_mpa_synth_init(mpa_window);
  41 }
  42
  43 /**
  44  * Process decoded Musepack data and produce PCM
  45  */
  46 static void mpc_synth(MPCContext *c, int16_t *out)
  47 {
  48     int dither_state = 0;
  49     int i, ch;
  50     OUT_INT samples[MPA_MAX_CHANNELS * MPA_FRAME_SIZE], *samples_ptr;
  51
  52     for(ch = 0;  ch < 2; ch++){
  53         samples_ptr = samples + ch;
  54         for(i = 0; i < SAMPLES_PER_BAND; i++) {
  55             ff_mpa_synth_filter(c->synth_buf[ch], &(c->synth_buf_offset[ch]),
  56                                 mpa_window, &dither_state,
  57                                 samples_ptr, 2,
  58                                 c->sb_samples[ch][i]);
  59             samples_ptr += 64;
  60         }
  61     }
  62     for(i = 0; i < MPC_FRAME_SIZE*2; i++)
  63         *out++=samples[i];
  64 }
  65
  66 void ff_mpc_dequantize_and_synth(MPCContext * c, int maxband, void *data)
  67 {
  68     int i, j, ch;
  69     Band *bands = c->bands;
  70     int off;
  71     float mul;
  72
  73     /* dequantize */
  74     memset(c->sb_samples, 0, sizeof(c->sb_samples));
  75     off = 0;
  76     for(i = 0; i <= maxband; i++, off += SAMPLES_PER_BAND){
  77         for(ch = 0; ch < 2; ch++){
  78             if(bands[i].res[ch]){
  79                 j = 0;
  80                 mul = mpc_CC[bands[i].res[ch]] * mpc_SCF[bands[i].scf_idx[ch][0]];
  81                 for(; j < 12; j++)
  82                     c->sb_samples[ch][j][i] = mul * c->Q[ch][j + off];
  83                 mul = mpc_CC[bands[i].res[ch]] * mpc_SCF[bands[i].scf_idx[ch][1]];
  84                 for(; j < 24; j++)
  85                     c->sb_samples[ch][j][i] = mul * c->Q[ch][j + off];
  86                 mul = mpc_CC[bands[i].res[ch]] * mpc_SCF[bands[i].scf_idx[ch][2]];
  87                 for(; j < 36; j++)
  88                     c->sb_samples[ch][j][i] = mul * c->Q[ch][j + off];
  89             }
  90         }
  91         if(bands[i].msf){
  92             int t1, t2;
  93             for(j = 0; j < SAMPLES_PER_BAND; j++){
  94                 t1 = c->sb_samples[0][j][i];
  95                 t2 = c->sb_samples[1][j][i];
  96                 c->sb_samples[0][j][i] = t1 + t2;
  97                 c->sb_samples[1][j][i] = t1 - t2;
  98             }
  99         }
 100     }
 101
 102     mpc_synth(c, data);
 103 }